Replace FINAL and OVERRIDE with their C++11 counterparts in content/renderer

content/renderer/media/webrtc_local_audio_track_unittest.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "base/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/mock_media_constraint_factory.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
@@ skipping 30 matching lines @@
     : capturer_(capturer),
       thread_(),
       closure_(false, false) {
     DCHECK(capturer);
     audio_bus_ = media::AudioBus::Create(params);
   }
 
   virtual ~FakeAudioThread() { DCHECK(thread_.is_null()); }
 
   // base::PlatformThread::Delegate:
-  virtual void ThreadMain() OVERRIDE {
+  virtual void ThreadMain() override {
     while (true) {
       if (closure_.IsSignaled())
         return;
 
       media::AudioCapturerSource::CaptureCallback* callback =
           static_cast<media::AudioCapturerSource::CaptureCallback*>(
               capturer_);
       audio_bus_->Zero();
       callback->Capture(audio_bus_.get(), 0, 0, false);
 
@@ skipping 29 matching lines @@
   MOCK_METHOD3(OnInitialize, void(const media::AudioParameters& params,
                                   CaptureCallback* callback,
                                   int session_id));
   MOCK_METHOD0(OnStart, void());
   MOCK_METHOD0(OnStop, void());
   MOCK_METHOD1(SetVolume, void(double volume));
   MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
 
   virtual void Initialize(const media::AudioParameters& params,
                           CaptureCallback* callback,
-                          int session_id) OVERRIDE {
+                          int session_id) override {
     DCHECK(params.IsValid());
     params_ = params;
     OnInitialize(params, callback, session_id);
   }
-  virtual void Start() OVERRIDE {
+  virtual void Start() override {
     audio_thread_.reset(new FakeAudioThread(capturer_, params_));
     audio_thread_->Start();
     OnStart();
   }
-  virtual void Stop() OVERRIDE {
+  virtual void Stop() override {
     audio_thread_->Stop();
     audio_thread_.reset();
     OnStop();
   }
  protected:
   virtual ~MockCapturerSource() {}
 
  private:
   scoped_ptr<FakeAudioThread> audio_thread_;
   WebRtcAudioCapturer* capturer_;
   media::AudioParameters params_;
 };
 
 // TODO(xians): Use MediaStreamAudioSink.
 class MockMediaStreamAudioSink : public PeerConnectionAudioSink {
  public:
   MockMediaStreamAudioSink() {}
   ~MockMediaStreamAudioSink() {}
   int OnData(const int16* audio_data,
              int sample_rate,
              int number_of_channels,
              int number_of_frames,
              const std::vector<int>& channels,
              int audio_delay_milliseconds,
              int current_volume,
              bool need_audio_processing,
-             bool key_pressed) OVERRIDE {
+             bool key_pressed) override {
     EXPECT_EQ(params_.sample_rate(), sample_rate);
     EXPECT_EQ(params_.channels(), number_of_channels);
     EXPECT_EQ(params_.frames_per_buffer(), number_of_frames);
     CaptureData(channels.size(),
                 audio_delay_milliseconds,
                 current_volume,
                 need_audio_processing,
                 key_pressed);
     return 0;
   }
@@ skipping 12 matching lines @@
   const media::AudioParameters& audio_params() const { return params_; }
 
  private:
   media::AudioParameters params_;
 };
 
 }  // namespace
 
 class WebRtcLocalAudioTrackTest : public ::testing::Test {
  protected:
-  virtual void SetUp() OVERRIDE {
+  virtual void SetUp() override {
     params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                   media::CHANNEL_LAYOUT_STEREO, 2, 48000, 16, 480);
     MockMediaConstraintFactory constraint_factory;
     blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio,
                              "dummy");
     MediaStreamAudioSource* audio_source = new MediaStreamAudioSource();
     blink_source_.setExtraData(audio_source);
 
     StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
                             std::string(), std::string());
     capturer_ = WebRtcAudioCapturer::CreateCapturer(
         -1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
         audio_source);
     audio_source->SetAudioCapturer(capturer_.get());
     capturer_source_ = new MockCapturerSource(capturer_.get());
     EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1))
         .WillOnce(Return());
     EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
     EXPECT_CALL(*capturer_source_.get(), OnStart());
     capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
   }
 
-  virtual void TearDown() OVERRIDE {
+  virtual void TearDown() override {
     blink_source_.reset();
     blink::WebHeap::collectAllGarbageForTesting();
   }
 
   media::AudioParameters params_;
   blink::WebMediaStreamSource blink_source_;
   scoped_refptr<MockCapturerSource> capturer_source_;
   scoped_refptr<WebRtcAudioCapturer> capturer_;
 };
 
@@ skipping 329 matching lines @@
   // Stopping the new source will stop the second track.
   EXPECT_CALL(*source.get(), OnStop()).Times(1);
   capturer->Stop();
 
   // Even though this test don't use |capturer_source_| it will be stopped
   // during teardown of the test harness.
   EXPECT_CALL(*capturer_source_.get(), OnStop());
 }
 
 }  // namespace content